search for: retransmitted

Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc32ed at 192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaaec88 at 192.168.1.95 for seqno 101

Hi, We have a Ericsson BP250 Phone system setup witht he following configuration Telco <-> Asterisk E405P <-> BP250 The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded. Currently running Asterisk 1.2.4 Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next

I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:02 WARNING[131081]: PRI:

Hi, What does retransmitting NAT means? I have a client that uses SPA 942, and his phone sometimes cannot be called. i did a sip sebug and i keep on seeing retransmitting NAT. on the realtime it shows that it is registered, so when i try to call it , asterisk thinks it is still online so it tries to reach it instead of saying it's unavailable, [Oct 9 11:10:33] -- Called 103100 it

...nd when running 30 channels) framing and "read" errors, with text similar to the following: WARNING[1167272128]: File chan_zap.c, Line 5670 (zt_pri_error): PRI: !! Got reject for frame 26, retransmitting frame 26 now, up_dating n_r! (repeating for each error several times, with ascending retransmitted frame numbers) and also, less often: WARNING[1167272128]: File chan_zap.c, Line 5670 (zt_pri_error): PRI: Read on NN failed: Unknown error 500 (NN is the channel) MY SETUP: Tyan S2723, with dual Xeon's running at 2.4 GHz, 1MB memory Redhat 9 Two TE410P's, spans set for E1. Problem happ...

Anyone know what would cause this error? !! Got reject for frame 39, retransmitting frame 39 now, updating n_r! !! Got reject for frame 39, retransmitting frame 40 now, updating n_r! I assume this would cause audio issues as well. Thanks, Steve > Message type: CALL PROCEEDING (2) > [18 03 a9 83 89] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0

I have an asterisk system in Costa Rica that connects to a SIP provider in Atlanta. Sometimes SIP packets seem get dropped or retransmitted too quickly. In trying to debug this I turned on SIP debug in Asterisk and the SIP provider enabled packet capture on his end. What I saw was me sending an invite, them sending a 100 Trying, me sending a cancel, me sending a retransmit of the cancel, me sending another retransmit of the cancel, t...

Hi, im trying to build a UAC and I'm coming up with some trouble whenever I receive a SIP 491 Request Pending Response. This happens because I try to place a call on hold using an Invite request rigth before Asterisk sends me a Re-Invite for the same call. I respond to the 491 response with an ACK however for some strange reason Asterisk doesn't accept the ACK and insists on retransmitting

I am running a Xen HVM guest with netfront PV drivers. This is running SLES10 SP3 inside the guest. The Dom0 is also SLES10 SP3. Now I am trying to communicate from that HVM guest to a Windows Visa or also Windows 7 machine and I am getting really poor TCP performance. When tracing on the network traffic, I can see that no packets are dropped or missing or anything, but what happens is that the

Hi all. I am having lots of trouble with random calls dropping after 20 seconds, and I finally managed to capture a full sip trace. I'll paste it in full below, but I'll give a summary first. It seems that Asterisk is not recognizing the ACK messages that it receives from the Grandstream ATA. This happens only on the ACK that follows the OK that marks a call as established. This makes

I agree with the in-effective of TCP transmission, but I wonder if the the UDP packet is dropped, the tinc VPN itself wouldn’t retransmit, and if the upper level application doesn’t handle the packet loss well, will this be the problem? Or the upper level application have very limited tolerance to packet loss(like RDP application, I guess if the packet loss go to certain threshold, the connection

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hi Everyone, I was hoping someone might know why I am experiencing a problem with Asterisk logging the event: [May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission 03f007af2b15cd0b54b0c368265d97be@sip.externalprovider.com for seqno 669371069 (Critical Response) This is happening after: - call is setup, 2 way audio - call can function correctly for up to 5

Hello everybody. I have a problem with an integration between an Asterisk (1.4.24.1) on FreeBSD 7.0 and a Shoretel 7.5 server. To make a very long story short, when someone behind asterisk call an extension behing shoretel everything work as expected. When someone behing the shoretel server call someone behind asterisk the first 10 seconds of the call seems ok but then the line is dropped

Hi, I'm currently investigating a problem with our windows 2008r2 guest on centos 6 hosts. The issue is that the windows system sometimes sees a SYN packet for a tcp connection but doesn't respond. Three seconds later the retransmitted packet arrives and this time windows decides to proceed normally with the connection. This is with the virtio drivers but I have now switched to the e1000 driver where the problem also occurs but not quite as badly as with virtio which leads me to speculate that the driver isn't really the...

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

Hi, I have set up the latest code * server, and I'm having problems with the sip phone. I'm using the budgetone. I can receive calls just fine. When I try to place outgoing calls, I can see that the rules get followed fine. But once it dials the analog interface (X100P), it starts retransmitting packets in chan_sip.c:retrans_pkt() any ideaas why this would be? thanks TJ

...5060-2 at 192.168.1.100 for seqno 11 (Critical Response) -- See doc/sip-retransmit.txt. [May 21 14:31:50] WARNING[25345]: chan_sip.c:2002 retrans_pkt: Hanging up call 954539948-5060-2 at 192.168.1.100 - no reply to our critical packet (see doc/sip-retransmit.txt). I read about SIP-packets being retransmitted but in the end not being acknowledged and so Asterisk ends the call. The network is like this : analogue phone -- Grandstream GXW4008 -- Linksys WAG160 -- Asterisk server -- ITSP Is this a NAT-problem in the router ?? Can it be resolved ? Kind regards, Jonas -------------- next part -----...

se?ores alguien le ha presentado este problema al acceder al voicemail o al hacer una llamada a la pstn 1940> Playing 'vm-received' (language 'es') -- <SIP/111-08d91940> Playing 'digits/yesterday' (language 'es') -- <SIP/111-08d91940> Playing 'digits/at' (language 'es') -- <SIP/111-08d91940> Playing

Hi All, I posted this a couple weeks ago with no response, I'm hoping that someone will see it this time around and be so kind as to offer advice for resolving this issue (or point me in the direction of a better place to ask) "Some" (but not all) calls on one of our Asterisk boxes are being dropped in Voicemail -- only in voicemail -- after about 20 seconds with the error logged